asterisk

  1. P

    Persistent Issues with SIM Card Blocks

    Hi everyone, We are currently facing a critical issue in our network of GSM gateways and SIM banks dedicated to VoIP GSM termination. Our SIM cards are unexpectedly ceasing to make calls, and we're seeking the community's expertise to help us troubleshoot and resolve this problem. Here's a...
  2. R

    Asterisk & Sip proxy problem

    Hello guys. i have a very urgent task to complete and i need help. I'm trying to use dsiprouter/kamailio as a sip proxy in front of my asterisk server. the sip endpoints are connected to dsiprouter and dsiprouter is configured as pass thru pbx and it forwards sip messages to my Asterisk server...
  3. M

    ARI-CLIENT SPEECH TO TEXT STREAMING

    Hi, Thank you for all the advices on the forum! But, I can't find the answer on what I need: I use ari-client in node-js (that is asked by my company) ari.connect(url, username, password, clientLoaded); async function clientLoaded (err, client) {if (err) {throw err;}...
  4. S

    Inbound TLS calls from Telnyx not disconnecting after hangup (PJSIP)

    I'm having a strange issue on inbound calls from Telnyx and I'm not sure if it's just my config, an issue with Telnyx or an issue with PJSIP. I have a SIP trunk on Telnyx that is configured for TLS/SRTP. Whenever I receive an inbound call, if I hangup the call on the PBX side, the call stays...
  5. G

    Asterisk Listen to incoming and outgoing calls

    Hello everyone someone masters asterisk please? I bought a SIP Trunk from OVH that I registered on the server in PJSIP. Absolutely everything works incoming calls and outgoing calls. I created a queue to make incoming calls wait with an audio output informing of the position of the caller for...
  6. J

    RCE Bugs in Hugely Popular VoIP Apps: Patch Now! (PJSIP RCE Vuln)

    https://threatpost.com/rce-bugs-popular-voip-apps-patch-now/178719/ FTA - 5 CVEs, now patched, affecting PJSUA. How to apply these patches on ipbx 2020 (debian 10)? Edit - on some further reading it seems (maybe) that PJSUA is not used in Asterisk's implementation of PJSIP? Idk enough...
  7. M

    QUESTION Asterisk - Outgoing call but I need to play an audio on another extension

    Let me try to explain, So, I have an softphone with the extension 5001 and an SIP device with an speaker on the extension 5000. I need to call some extension/context and instead on hearing the audio myself (On a call from extension 5001 to the extension 5000), I need that this audio plays on...
  8. Rodrigo Cuadra

    TUTORIAL Implement WebRTC in Asterisk

    As part of contributing to the Asterisk community, VitalPBX has just published a new Blog on how to implement WebRTC with PJSIP in Asterisk from scratch. We invite you to read our new Blog which is in English and Spanish. English Version...
  9. AvalonWien

    update callerid on attended transfer

    A call comes in. I pick up at my extension (111) and have the number of the external caller (012345678) in my display. Perfect. The call is not for me, so I place a new call to extension 222. The calle (222) sees my extension (111) in his display. Perfect. But when I transfer the incoming call...
  10. J

    TIPS I need help

    Hi, firstly sorry for my English . i need urgently my asterisk ready for 3 days but I don’t know anything about that Can you give me the best tutorials and recents please with -installation and the guide for spoof a caller id. and I see in the forum I need an other software for call can you...
  11. jbrandon.cj

    Asterisk detects ipv6 and doesn't play audio

    Specs: CentOS 7, Incredible PBX 2020 Machine, Fresh Install. IPv6 is disabled within CentOS. This is a public facing pbx and the public script has been ran. Some sip clients are only being detected by their ipv6 address and not the ipv4 and might not play audio. Everything works fine over...
  12. S

    Asterisk + Flexisip + Push Notification

    Hi i've MirtaPBX with Asterisk and i want to use Flexisip as Push notification so the reason i want to use flexisip is, the Android and IOS Phones client (Customized Linphone) can be awaken whenever their configured and registered extension gets a call. Im sure you may all know that SIP client...
  13. A

    SCCP Phone cannot receive calls

    Hello everyone, I have a problem completing the setup of a Cisco phone. This is what I did: Set up a Raspbx Configured 3 SIP softphones. Added a Cisco 6901 SCCP phone as following: Installed SCCP module with: apt-get update apt-get install chan-sccp-ast16 Set pbx ip address as t*f*t*p server...
  14. S

    MiddleEast SIP Block - Remote Worker Solution Needed

    Hi, so I have a friend in the Middle East and in his country SIP is blocked and prohibited. Also, ISPs are not allowed to give Public IP neither they do port forwarding in their Internet Router He has incrediablepbx integrated with GoIP (SIM Gateways) and a couple of Astra Phones in his office...
  15. C

    Dialing late-registering endpoints PJSIP channel

    Hi all, I was looking for the most efficient way to Dial late-registering SIP endpoints on the PJSIP channel. I was assuming to use RetryDial for this, however this has following problems: It does not start calling when no PJSIP destination endpoint is registered at the moment the call...
  16. M

    Questions about Asterisk features and other generalities

    Hello, (warning : I'm a complete beginner) 1) Is there an alternative (in parallel) to the phone number for calls between asterisk servers ? Like calling a hostname:port ? (without configuration modification for each new server) 2) When using a trunking sip service, can this service...
  17. L

    Incoming Calls are working but no outgoing calls!__Newbie_HelpRequired

    I have a SIP Trunk configured and is registered in FreePBX. The SIP is registered and incoming calls working properly with incoming routes to an extension whereas the outbound calls are failing with All Circuits are busy now, please try your call later; prompt. Then the asterisk -r screen shows...
  18. A

    SOLVED Cisco 7970 not registering with Asterisk

    Hello I'm having problems registering my SIP configured 7970 phone with my Asterisk server. The phone is using SIP 70.9-2-1S firmware, and my SEPmac.conf.xml file looks like this: <?xml version="1.0" encoding="UTF-8"?> <device> <deviceProtocol>SIP</deviceProtocol>...
  19. P

    Need urgent help

    Hello all, I have gotten a task in school regarding Asterisk, I have tried going through the documentation to see what the code that I will write bellow does, but I honestly cant put it together. Can anyone please try to explain what does it do. It will mean a lot. This is a Dialplan...
  20. B

    PIONEERS Sparrow: Run Wazo engine on Raspberry PI

    Hello everyone, I m happy to introduce the first release of Sparow. Sparrow is an unofficial build for armhf architecture of Wazo engine. This allows you to run a Wazo engine on a Raspberry Pi. This first version is based on version 19.17 of Wazo. The project website is available at this...
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